This invention relates to methods for digital compression and digital expansion of linear based or sine-based signals (e.g. audio or other signal types) by determination of a signal coordination point in a bi-adaptive scaleable mV/step and a time/step structured plane.
The general principle of digital signal data flow will be described with reference to FIG. 1. Transporting the input signal information via satellite, or storing the input signal information in memory, requires an “analogue,” or analog, signal source c1 (analog transducer signal, e.g. output of an analog strain measurement or an analog sine-based signal source (e.g. audio from a microphone output), which will be transferred at d1 to the signal coder c2 (signal digitizing, compression) and back to a signal decoder c3 (signal decompression and analogizing), and to an analog signal output c4 (e.g. fed to an amplifier and a loudspeaker—not shown).
For all applications, it is important to transfer a maximum signal quality at a minimum data rate.
An object of this invention is to create a method for compression and expansion of linear based analog signals (e.g. non-audio based signals) or sine-based analog signals (e.g. audio based signals) that provide a minimal loss of signal characteristics at a very low data rate.
This object is achieved by way of a method for digital compression and digital expansion of linear or sine-based signals by determination of a signal coordination point in a bi-adaptive scaleable mV/step and a time/step structured plane. The method includes digitizing an analog input signal, detecting breaks of the digitized input signal, and determining a time difference and an amplitude difference of two successive breaks of the input signal. The time difference and the amplitude difference of successive breaks are value coded as a data word based on adaptive scaleable time-per-step tables and voltage-per-step tables, and the time-per-step tables and the voltage-per-step tables are selected depending on an absolute value of the time difference and amplitude difference determined so as to produce compressed data. Preferred embodiments of the invention, as well as a corresponding expansion method, are also claimed.
In an input signal compression method according to the invention, the input signal is digitized via an A/D converter, the breaks (maximum values or kinks in the signal) of the digitized input signal are detected, and the time difference and the amplitude difference of two successive breaks of the input signal are determined.
The time difference and the amplitude difference of successive breaks are value coded as a data word on the basis of scaleable time-per-step tables and scaleable voltage-per-step tables, with the time-per-step tables and the voltage-per-step tables being selectable depending on the absolute break position differences in the mV/step and time/step structured planes, resulting from the determined time differences and amplitude differences of the detected input signal breaks.
Thus, by using adaptive scaleable tables, depending on the time difference and associated amplitude difference, it is possible to build a two dimensional time-per-step and voltage-per-step structured plane for every successive break position of the input signal. Based on this procedure, the data rate of the input signal coding process can be dynamically adapted to the input signal frequency and the signal amplitude for every break-to-break distance.
As a consequence, the necessary memory for storing the compressed audio data will decrease. On the other hand, the input signal recording time at a given memory size will increase.
By way of the invention, it is possible to transfer mechanical sourced signals (linear based signals) that are particularly relevant to mechanical defect investigation of industrial machines (e.g. turbines, gears, analog sensors) as well as human vocal-based audio (sine-based signals).
These and other objects, aspects and embodiments of the present invention will be described in more detail with reference to the drawings.